1. Technical Field
The present methods and systems relate to media communications such as voice or video over packet-switched networks and, more particularly, to efficient methods and systems for handover of active media sessions when mobile devices move between heterogeneous Internet Protocol (IP) networks.
2. Description of Related Art
The use of public packet-switched networks for voice and video communications is expected to continue to rapidly grow. “Internet telephony” is one example of packet-switched telephony. In packet-switched telephony, a packet-switched network such as the Internet serves as a transportation medium for packets carrying data for voice, video, or other media such as music. Voice-over-Internet-Protocol (VoIP) is one example of a collection of standards and protocols used to support voice or video communications over packet-switched networks such as the Internet. Others have been developed as well, including proprietary protocols such as Skype®. A common Internet telephony scheme involves a computer or other device that is capable of connecting to the Internet, transmitting voice or video to a second device such as a gateway, another mobile device, or another computer.
A long-term, planned evolution of existing mobile networks worldwide is to migrate the underlying mobile network protocols and systems from a traditional circuit-switched architecture for voice services to a packet switched architecture that is based on IP standards. Prominent examples include the 3GPP (3rd Generation Partnership Project) consortium's announced “Long-Term Evolution” (LTE) standards as well as the Institute of Electrical and Electronic Engineers' (IEEE) mobile WiMax standards (802.16 or 802.20) and subsequent versions. Although these standards continue to evolve, a common expected feature is that the underlying networks will remain packet switched and based on IP standards such as IPv4 and IPv6.
With these next-generation networks, voice, video, or other media is transmitted over the IP network with access to the public Internet provided by the mobile operator or wireless Internet Service Provider (ISP). Media sessions such as voice calls can be managed by protocols such as the Session Initiation Protocol (SIP) or Extensible Messaging and Presence Protocol (XMPP), and these protocols generally operate at the application layer of the Open Systems Interconnection (OSI) stack. The IP Multimedia Subsystem (IMS) as specified by 3GPP is another example of next-generation mobile-network technology that implements voice calls and media sessions via application-layer software.
Numerous benefits may be realized through the use of packet-switched telephony based on IP standards for mobile networks. First, mobile operators would like to provide access data services such as web-browsing, e-mail, or video through the mobile device, and these services are frequently provided through standard Internet protocols such as Hypertext Transfer Protocol (HTTP) for web browsing, Simple Mail Transfer Protocol (SMTP) for e-mail, or Real-Time Streaming Protocol (RTSP) for video. Thus, mobile network operators may efficiently support many data services through the deployment of an IP network and by assigning IP addresses to mobile devices. With an efficient IP network, the traditional circuit-switched network becomes redundant, and the mobile operator can lower costs and simplify operations by managing a single IP network instead of both an IP network and a circuit-switched network.
Further, voice services worldwide are migrating to IP networks, and the International Telecommunications Union (ITU) projected that nearly 50% of all international phone calls were carried by VoIP in 2007. As the corresponding endpoints for telephone calls or media sessions are increasingly located on IP networks, the most efficient method for connecting calls should be to keep the call control and media on the Internet when possible, as opposed to (i) transferring calls to a circuit-switched network such as the traditional Public Switched Telephone Network (PSTN) and then (ii) transferring the call back to the Internet in order to reach another endpoint that is also connected to the Internet. Also, keeping the VoIP telephone calls from mobile devices natively on the Internet can reduce costs, since traditional per-minute charges for connecting phone calls with the legacy PSTN can be bypassed.
Although the implementation of an underlying IP network for all mobile services provides the significant benefits noted above, several important new challenges are created. An important class of problems relates to handover of active media sessions such as voice calls when the mobile device moves through the mobile network or between different networks. A significant focus of the various wireless Wide Area Network (WAN) standards such as 802.11e and 802.20 pertains to keeping an IP address bound to the mobile device or Media Access Control (MAC) address, even if the mobile device changes base stations due to movement such as driving in a car.
Traditional VoIP protocols such as SIP, Extensible Messaging and Presence Protocol (XMPP), H.323, Media Gateway Control Protocol (MCGP), or Inter-Asterisk Exchange (IAX2), as well as current implementations of proprietary protocols such as Skype®, were originally designed when almost all devices with IP addresses could reasonably be expected to keep the same IP address for the duration of a media session. Further, active sessions in many applications that implement transmission control protocol (TCP) were also not designed for the IP address to change during the session, such as Secure Shell (SSH) or File Transfer Protocol (FTP). Consequently, mobile operators and users may prefer for the IP address of the mobile device to not change during active sessions, and next generation mobile networks using protocols such as WiMax and LTE are generally designed to minimize the change of IP addresses assigned to a given device.
For voice services, changing the IP address during an active call can result in noticeable gaps or delay in the audio for both sides, especially when the IP address is changed as the subscriber moves between completely different networks, such as from a macro-cellular WiMax network to a WiFi (802.11) network in a building. Such a change in IP addresses for the mobile device could be a likely scenario when the subscriber walks into an office building from the street, for example, where a superior connection to the Internet is provided through a local WiFi connection instead of a wide area WiMax connection.
What is needed in the art are techniques for seamless handover of an active telephone call when the preferred IP address of the mobile device changes, such that potential gaps or distortion of audio are minimized, in order to efficiently maintain “peer-to-peer” communication with a corresponding node. What is also needed in the art is the avoidance of a disconnected call upon an IP address change, which would require the user to place a second call to continue the conversation. When the Internet connectivity for the example WiFi network is provided by an ISP that is a different entity from the mobile operator, the two subnets of the public Internet are commonly referred to as “heterogeneous” networks in that they may be separately-managed subnets of the public Internet.
Under the above-referenced scenario of handing over an active media session such as a voice call when a mobile device acquires a new, preferred IP address for communication, a need in the art exists to solve the significant class of problems introduced by various types of Network Address Translation (NAT) routers with port translation and/or firewalls that may operated between a mobile device and a corresponding node (such as a gateway, a packet-based telephone, a personal computer, another mobile device, some other endpoint, etc.). A need exists in the art to solve the problems for handover created by port translation, firewalls which may block either inbound or outbound packets, and application layer gateways (ALGs) that may either alter packets or block packets. The handover of an active call between heterogeneous networks is also referred to as “vertical handover”. For example, an ALG, managed by the mobile operator and located between the mobile network and the public Internet, may both (i) compress IP headers and also (ii) perform NAT routing functions.
The corresponding node may also be behind (i.e. be situated on the other side of the public Internet from) a NAT router or firewall, or possibly multiple levels of NAT routers and firewalls. Many NAT routers translate ports in addition to translating IP addresses, thereby allowing multiple hosts within an internal network to share a single public IP address. In order to reduce the potential gaps or delays in the audio during transfer of an existing media session to a new IP address, a need exists to handover an active media session by properly addressing the complexities of intermediate devices that either (i) alter the IP packets transmitted and received by either a mobile device or a corresponding node or (ii) drop packets that do not match pre-determined firewall rules.
There is a need in the art to address the scenario in which the WiFi network and the corresponding node are both behind NAT routers (or “NATs” for short). The use of NATs is expected to increase significantly as the number of broadband connections worldwide increases while (i) the available pool of public IPv4 addresses remains fixed and (ii) the transition from IPv4 to IPv6 proceeds relatively slowly. For example, the CTO of NTT America estimated that there were only 2,000 consumers in Japan accessing the Internet via IPv6 connections, five years after the introduction of IPv6 services by NTT in Japan (Network World, Mar. 31, 2008). IMS Research projected that the number of fixed broadband connections globally will grow from 150 million in 2005 to over 400 million by 2009. There is also a need in the art for vertical handovers to support Internet standards that are actually widely deployed on common consumer and business NAT routers and firewalls, such as utilizing handover techniques primarily through User Datagram Protocol (UDP) or TCP standards.
Although applications such as VoIP or web browsing operating on a mobile device may prefer to keep the same IP address active during a session under normal circumstances, there are many instances when routing the packets through a new or different IP address on the mobile device (MD) may be preferred in order to provide the superior service to the subscriber. A mobile device that provides services through Internet protocols will generally prefer a link-layer connection that provides higher signal-to-noise ratios, lower power requirements, lower costs for the IP connectivity, and/or other benefits. When the mobile device is in the proximity of an 802.11 access point, connecting to the Internet through the 802.11 access point instead of the macro-cellular network may be preferred.
In addition, WiFi technology is currently widely deployed, and the number of access points globally is expected to continue to increase. For example, the market research firm In-Stat estimated that 213 million Wi-Fi chipsets were shipped out worldwide in 2006, representing a 32% growth rate over 2005. When a subscriber places a telephone call on a mobile device that connects to the Internet through WiFi, such as at the subscriber's home or office, and then the subscriber moves out of WiFi range by leaving the premises, and the macro-cellular network begins providing IP connectivity, the underlying or locally assigned IP address of the mobile device will likely change. There could also likely be a period of time when both the IP address from the WiFi network and the IP address from the wireless wide area network (WAN) are available and active.
Similarly, superior connectivity to the mobile device may be delivered by changing from the macro-cellular network to WiFi. For example, if a macro-cellular WiMax network is provided through the 2.50-2.69 GHz frequency bands identified in the ITU WRC-2000 recommendations, the macro-cellular network signal may be degraded by building walls and an improved connection for the device could be obtained by switching the IP connectivity for the mobile device from the macro-cellular network to WiFi when a subscriber walks inside a building with WiFi access. In some instances, the quality of the macro-cellular connection could degrade to the point where it is unusable while WiFi connectivity is strong, such as if the subscriber moves into the basement of a building in close proximity to a WiFi base station. In this example scenario, there exists a need in the art to change the IP address that the mobile device utilizes for communicating with a corresponding node as rapidly as possible through minimizing requirements for call-control signaling, as the preferred routing of both inbound and outbound packets is changed from the macro network to WiFi.
There may be a period of time when both IP addresses are simultaneously active and available to applications on the mobile device, and a need exists in the art to (i) support “make before break” handover, while (ii) simultaneously addressing the significant need existing in the art to address the complexities of NAT routers and/or firewalls that may be located in front of either or both endpoints during handover of a media session. There exists a need in the art to support handover between WiFi and macro-cellular networks that implement VoIP for telephone calls in order to provide significant benefits to the mobile operator and subscriber through increased network coverage. There is also a need in the art for efficient handover techniques that will reduce the potential for dropped media packets or increased jitter in a manner that requires minimal additional servers or processes for the mobile network or communications service provider to manage. A need exists in the art to perform the handover in a rapid manner.
The IP address of the mobile device may also change when the subscriber moves between separate mobile operator networks that both provide IP connectivity, analogous to roaming in traditional 2G and 3G mobile networks. A need exists in the art to properly execute the handover when the preferred IP address changes, thus allowing an “all IP” network infrastructure to keep active telephone calls from being disconnected, even though the subscriber moves between completely-separate networks. In legacy mobile networks such as those with GSM 2G technology, a subscriber's telephone call will generally not remain active if, for example, a T-Mobile® subscriber moves from a location serviced by T-Mobile® to a location serviced by AT&T® (and not by T-Mobile®), even though AT&T® and T-Mobile® may have roaming agreements that allow idle handovers.
The handover of active calls between heterogeneous GSM 2G networks may not be commonly supported because the roaming agreements and the implementation of GSM protocols may not support roaming where calls stay active even though the subscriber moves to a completely-separate network. In contrast, the WiMax specification generally assumes voice and other media services are managed at the application layer of the traditional OSI stack. For example, two separate networks such as Clearwire® and NextWave® may provide mobile services through WiMax. If a subscriber belonging to the Clearwire® network moves from a location serviced by Clearwire® to a location serviced by NextWave® (but not Clearwire®), the IP address of the mobile device will likely change. There exists a need in the art for seamless handover of an active telephone call or other media sessions, even though the subscriber has moved between the two heterogeneous IP-based mobile networks, also known as making a vertical handover.
A need exists in the art for the vertical handover to be managed at the application layer of the OSI stack. Next-generation mobile networks generally assume voice services are transmitted through VoIP and managed at the application layer. This network architecture provides significant opportunities for companies managed independently of the mobile network operators, such as Skype®, Google®, Yahoo®, or Truphone® to provide voice, video, or other media services. These and similar companies may offer end users a communications service. The communications service could be delivered through software programs operating on a mobile device and may have access to the Internet via the mobile operator's data network.
However, the software programs may not have control over low-level functions such as managing the MAC address or associating the mobile device with a particular base station. What is needed in the art are efficient methods and systems to support active call handover for communications services, as the mobile device moves between heterogeneous IP networks. There exists a need in the art for software programs to (i) detect new IP addresses becoming available on the mobile device and (ii) monitor the quality of the links to decide that a handover is desirable, and (iii) execute the handover as efficiently as possible. Thus, there exists a need in the art for seamless handover to be managed via software operating as an application on the mobile device.
In addition, a need exists in the art to, in the event of handover, adequately support media control protocols, such as the Real-Time Control Protocol (RTCP) or Secure Real-Time Control Protocol (SRTCP), that may operate on a separate port than that used for media. Media control protocols can be helpful for monitoring quality and reporting back to the originating device of a media stream about the attributes of media received at another node. A need exists to provide feedback to a node through media control channels, allowing the node to, for example, introduce additional channel coding in the media if excessive bit errors are observed on the receiving side.
A further need exists, in the case where packet loss is observed on the terminating side, to provide feedback to the originating device in order to implement forward error correction (FEC) codes such as packet duplication, or switch to a different, more frame-independent codec such as switching from G.729b to the Internet Low-Bandwidth Codec (iLBC). A need exists in the art to efficiently handover media control messages from a receiving device to a transmitting device in order to quickly evaluate quality of media during handover and determine if the handover is successful or complete.
Further, a need exists in the art to provide direct communication between a mobile device and a corresponding node, in order to provide a more efficient handover of media than is provided by alternative techniques such as Mobile IP (IETF RFC 3775 and 4721, which are hereby incorporated herein by reference). Notable limitations of Mobile IP include triangular routing of media packets between a mobile device and a corresponding node, the complexity and cost of managing servers such as Home Agents for media sessions, implementing virtual Network Interface Cards (NICs) on a mobile device, and multiple other requirements and signaling overhead as specified in the Mobile IP standards. And other needs exist in the art as well, as the list recited above is not meant to be exhaustive but rather illustrative.